A set of raw (decoded) samples, plus a timestamp for when to play those samples relative to other items in a given IContainer. More...

#include <IAudioSamples.h>

Inheritance diagram for com::avpkit::core::IAudioSamples:

[legend]

Collaboration diagram for com::avpkit::core::IAudioSamples:

[legend]

Public Types
enum	Format { FMT_NONE = -1 , FMT_U8 , FMT_S16 , FMT_S32 , FMT_FLT , FMT_DBL , FMT_U8P , FMT_S16P , FMT_S32P , FMT_FLTP , FMT_DBLP }
	The format we use to represent audio. More...

enum class	ChannelLayout : uint64_t { CH_NONE = 0 , CH_FRONT_LEFT = 0x00000001 , CH_FRONT_RIGHT = 0x00000002 , CH_FRONT_CENTER = 0x00000004 , CH_LOW_FREQUENCY = 0x00000008 , CH_BACK_LEFT = 0x00000010 , CH_BACK_RIGHT = 0x00000020 , CH_FRONT_LEFT_OF_CENTER = 0x00000040 , CH_FRONT_RIGHT_OF_CENTER = 0x00000080 , CH_BACK_CENTER = 0x00000100 , CH_SIDE_LEFT = 0x00000200 , CH_SIDE_RIGHT = 0x00000400 , CH_TOP_CENTER = 0x00000800 , CH_TOP_FRONT_LEFT = 0x00001000 , CH_TOP_FRONT_CENTER = 0x00002000 , CH_TOP_FRONT_RIGHT = 0x00004000 , CH_TOP_BACK_LEFT = 0x00008000 , CH_TOP_BACK_CENTER = 0x00010000 , CH_TOP_BACK_RIGHT = 0x00020000 , CH_STEREO_LEFT = 0x20000000 , CH_STEREO_RIGHT = 0x40000000 , CH_WIDE_LEFT = 0x0000000080000000ULL , CH_WIDE_RIGHT = 0x0000000100000000ULL , CH_SURROUND_DIRECT_LEFT = 0x0000000200000000ULL , CH_SURROUND_DIRECT_RIGHT = 0x0000000400000000ULL , CH_LOW_FREQUENCY_2 = 0x0000000800000000ULL , CH_LAYOUT_NATIVE = 0x8000000000000000ULL , , channel_mask_c , CH_LAYOUT_MONO = (CH_FRONT_CENTER) , CH_LAYOUT_STEREO = (CH_FRONT_LEFT\|CH_FRONT_RIGHT) , CH_LAYOUT_2POINT1 = (CH_LAYOUT_STEREO\|CH_LOW_FREQUENCY) , CH_LAYOUT_2_1 = (CH_LAYOUT_STEREO\|CH_BACK_CENTER) , CH_LAYOUT_SURROUND = (CH_LAYOUT_STEREO\|CH_FRONT_CENTER) , CH_LAYOUT_3POINT1 = (CH_LAYOUT_SURROUND\|CH_LOW_FREQUENCY) , CH_LAYOUT_4POINT0 = (CH_LAYOUT_SURROUND\|CH_BACK_CENTER) , CH_LAYOUT_4POINT1 = (CH_LAYOUT_4POINT0\|CH_LOW_FREQUENCY) , CH_LAYOUT_2_2 = (CH_LAYOUT_STEREO\|CH_SIDE_LEFT\|CH_SIDE_RIGHT) , CH_LAYOUT_QUAD = (CH_LAYOUT_STEREO\|CH_BACK_LEFT\|CH_BACK_RIGHT) , CH_LAYOUT_5POINT0 = (CH_LAYOUT_SURROUND\|CH_SIDE_LEFT\|CH_SIDE_RIGHT) , CH_LAYOUT_5POINT1 = (CH_LAYOUT_5POINT0\|CH_LOW_FREQUENCY) , CH_LAYOUT_5POINT0_BACK = (CH_LAYOUT_SURROUND\|CH_BACK_LEFT\|CH_BACK_RIGHT) , CH_LAYOUT_5POINT1_BACK = (CH_LAYOUT_5POINT0_BACK\|CH_LOW_FREQUENCY) , CH_LAYOUT_6POINT0 = (CH_LAYOUT_5POINT0\|CH_BACK_CENTER) , CH_LAYOUT_6POINT0_FRONT = (CH_LAYOUT_2_2\|CH_FRONT_LEFT_OF_CENTER\|CH_FRONT_RIGHT_OF_CENTER) , CH_LAYOUT_HEXAGONAL = (CH_LAYOUT_5POINT0_BACK\|CH_BACK_CENTER) , CH_LAYOUT_6POINT1 = (CH_LAYOUT_5POINT1\|CH_BACK_CENTER) , CH_LAYOUT_6POINT1_BACK = (CH_LAYOUT_5POINT1_BACK\|CH_BACK_CENTER) , CH_LAYOUT_6POINT1_FRONT = (CH_LAYOUT_6POINT0_FRONT\|CH_LOW_FREQUENCY) , CH_LAYOUT_7POINT0 = (CH_LAYOUT_5POINT0\|CH_BACK_LEFT\|CH_BACK_RIGHT) , CH_LAYOUT_7POINT0_FRONT = (CH_LAYOUT_5POINT0\|CH_FRONT_LEFT_OF_CENTER\|CH_FRONT_RIGHT_OF_CENTER) , CH_LAYOUT_7POINT1 = (CH_LAYOUT_5POINT1\|CH_BACK_LEFT\|CH_BACK_RIGHT) , CH_LAYOUT_7POINT1_WIDE = (CH_LAYOUT_5POINT1\|CH_FRONT_LEFT_OF_CENTER\|CH_FRONT_RIGHT_OF_CENTER) , CH_LAYOUT_7POINT1_WIDE_BACK = (CH_LAYOUT_5POINT1_BACK\|CH_FRONT_LEFT_OF_CENTER\|CH_FRONT_RIGHT_OF_CENTER) , CH_LAYOUT_OCTAGONAL = (CH_LAYOUT_5POINT0\|CH_BACK_LEFT\|CH_BACK_CENTER\|CH_BACK_RIGHT) , CH_LAYOUT_HEXADECAGONAL = (CH_LAYOUT_OCTAGONAL\|CH_WIDE_LEFT\|CH_WIDE_RIGHT\|CH_TOP_BACK_LEFT\|CH_TOP_BACK_RIGHT\|CH_TOP_BACK_CENTER\|CH_TOP_FRONT_CENTER\|CH_TOP_FRONT_LEFT\|CH_TOP_FRONT_RIGHT) , CH_LAYOUT_STEREO_DOWNMIX = (CH_STEREO_LEFT\|CH_STEREO_RIGHT) }

typedef enum com::avpkit::core::IAudioSamples::Format	Format
	The format we use to represent audio. More...

Public Member Functions
virtual bool	isComplete ()=0
	Returns whether or not we think this buffer has been filled with data. More...

virtual int32_t	getSampleRate ()=0
	Find the sample rate of the samples in this audio buffer. More...

virtual int32_t	getChannels ()=0
	Return the number of channels of the samples in this buffer. More...

virtual int32_t	getSampleBitDepth ()=0
	Find out the bit-depth of the samples in this buffer. More...

virtual Format	getFormat ()=0
	Find the Format of the samples in this buffer. More...

virtual int32_t	getNumSamples ()=0
	Get the number of samples in this video. More...

virtual int32_t	getMaxBufferSize ()=0

virtual int32_t	getMaxSamples ()=0

virtual int32_t	getSampleSize ()=0

virtual int64_t	getPts ()=0
	What is the Presentation Time Stamp of this set of audio samples. More...

virtual void	setPts (int64_t aValue)=0
	Set the Presentation Time Stamp for this set of samples. More...

virtual int64_t	getNextPts ()=0
	What would be the next Presentation Time Stamp after all the samples in this buffer were played? More...

virtual void	setComplete (bool complete, int32_t numSamples, int32_t sampleRate, int32_t channels, Format format, int64_t pts)=0
	Call this if you modify the samples and are now done. More...

virtual short *	getRawSamples (uint32_t startingSample)=0

virtual int32_t	setSample (int32_t sampleIndex, int32_t channel, Format format, int32_t sample)=0
	Sets the sample at the given index and channel to the sample. More...

virtual int32_t	getSample (int32_t sampleIndex, int32_t channel, Format format)=0
	Get the sample at the given sampleIndex and channel, and return it in the asked for format. More...

virtual ChannelLayout	getChannelLayout ()=0

virtual void	setComplete (bool complete, int32_t numSamples, int32_t sampleRate, int32_t channels, ChannelLayout channelLayout, Format format, int64_t pts)=0

Public Member Functions inherited from com::avpkit::core::IMediaData
virtual int64_t	getTimeStamp ()=0
	Get the time stamp of this object in getTimeBase() units. More...

virtual void	setTimeStamp (int64_t aTimeStamp)=0
	Set the time stamp for this object in getTimeBase() units. More...

virtual IRational *	getTimeBase ()=0
	Get the time base that time stamps of this object are represented in. More...

virtual void	setTimeBase (IRational *aBase)=0
	Set the time base that time stamps of this object are represented in. More...

virtual com::avpkit::ferry::IBuffer *	getData ()=0
	Get any underlying raw data available for this object. More...

virtual int32_t	getSize ()=0
	Get the size in bytes of the raw data available for this object. More...

virtual bool	isKey ()=0
	Is this object a key object? i.e. More...

virtual void	setData (com::avpkit::ferry::IBuffer *buffer)=0
	Sets the underlying buffer used by this object. More...

Public Member Functions inherited from com::avpkit::ferry::RefCounted
virtual int32_t	acquire ()
	Internal Only. More...

virtual int32_t	release ()
	Internal Only. More...

virtual RefCounted *	copyReference ()
	Create a new Java object that refers to the same native object. More...

virtual int32_t	getCurrentRefCount ()
	Return the current reference count on this object. More...

void	setJavaAllocator (void *allocator)
	This method is public but not part of the standard API. More...

void *	getJavaAllocator ()
	This method is public but not part of the standard API. More...

Static Public Member Functions
static int32_t	findSampleBitDepth (Format format)
	A convenience method that returns the # of bits in a given format. More...

static IAudioSamples *	make (int32_t numSamples, int32_t numChannels)
	Get a new audio samples buffer. More...

static int64_t	samplesToDefaultPts (int64_t samples, int sampleRate)
	Converts a number of samples at a given sampleRate into Microseconds. More...

static int64_t	defaultPtsToSamples (int64_t duration, int sampleRate)
	Converts a duration in microseconds into a number of samples, assuming a given sampleRate. More...

static IAudioSamples *	make (com::avpkit::ferry::IBuffer *buffer, int channels, IAudioSamples::Format format)
	Creates an IAudioSamples object by wrapping an object. More...

static IAudioSamples *	make (int32_t numSamples, int32_t numChannels, IAudioSamples::Format format)
	Get a new audio samples buffer. More...

Additional Inherited Members
Protected Member Functions inherited from com::avpkit::ferry::RefCounted
virtual void	destroy ()
	This method is called by RefCounted objects when their Ref Count reaches zero and they are about to be destroyed.

Protected Attributes inherited from com::avpkit::ferry::RefCounted
AtomicInteger *	mRefCount
	This is the internal reference count, represented as an AtomicInteger to make sure it is thread safe.

void *	mAllocator
	Not part of public API.

Detailed Description

A set of raw (decoded) samples, plus a timestamp for when to play those samples relative to other items in a given IContainer.

The timestamp value in decoded data is always in Microseonds.

Definition at line 37 of file IAudioSamples.h.

Member Typedef Documentation

◆ Format

typedef enum com::avpkit::core::IAudioSamples::Format com::avpkit::core::IAudioSamples::Format

The format we use to represent audio.

Today only FMT_S16 (signed integer 16-bit audio) is supported.

Member Enumeration Documentation

◆ ChannelLayout

enum com::avpkit::core::IAudioSamples::ChannelLayout : uint64_t

strong

Enumerator
CH_STEREO_LEFT	Stereo downmix.
CH_STEREO_RIGHT	See CH_STEREO_LEFT.
CH_LAYOUT_NATIVE	Channel mask value used for AVCodecContext.request_channel_layout to indicate that the user requests the channel order of the decoder output to be the native codec channel order.

Definition at line 276 of file IAudioSamples.h.

                              : uint64_t {
 CH_NONE=                   0,        
 CH_FRONT_LEFT=             0x00000001,
 CH_FRONT_RIGHT=            0x00000002,
 CH_FRONT_CENTER=           0x00000004,
 CH_LOW_FREQUENCY=          0x00000008,
 CH_BACK_LEFT=              0x00000010,
 CH_BACK_RIGHT=             0x00000020,
 CH_FRONT_LEFT_OF_CENTER=   0x00000040,
 CH_FRONT_RIGHT_OF_CENTER=  0x00000080,
 CH_BACK_CENTER=            0x00000100,
 CH_SIDE_LEFT=              0x00000200,
 CH_SIDE_RIGHT=             0x00000400,
 CH_TOP_CENTER=             0x00000800,
 CH_TOP_FRONT_LEFT=         0x00001000,
 CH_TOP_FRONT_CENTER=       0x00002000,
 CH_TOP_FRONT_RIGHT=        0x00004000,
 CH_TOP_BACK_LEFT=          0x00008000,
 CH_TOP_BACK_CENTER=        0x00010000,
 CH_TOP_BACK_RIGHT=         0x00020000,
 CH_STEREO_LEFT=            0x20000000,  
 CH_STEREO_RIGHT=           0x40000000,  
 CH_WIDE_LEFT=              0x0000000080000000ULL,
 CH_WIDE_RIGHT=             0x0000000100000000ULL,
 CH_SURROUND_DIRECT_LEFT=   0x0000000200000000ULL,
 CH_SURROUND_DIRECT_RIGHT=  0x0000000400000000ULL,
 CH_LOW_FREQUENCY_2=        0x0000000800000000ULL,
  
 CH_LAYOUT_NATIVE=          0x8000000000000000ULL,
  
 CH_LAYOUT_MONO=              (CH_FRONT_CENTER),
 CH_LAYOUT_STEREO=            (CH_FRONT_LEFT|CH_FRONT_RIGHT),
 CH_LAYOUT_2POINT1=           (CH_LAYOUT_STEREO|CH_LOW_FREQUENCY),
 CH_LAYOUT_2_1=               (CH_LAYOUT_STEREO|CH_BACK_CENTER),
 CH_LAYOUT_SURROUND=          (CH_LAYOUT_STEREO|CH_FRONT_CENTER),
 CH_LAYOUT_3POINT1=           (CH_LAYOUT_SURROUND|CH_LOW_FREQUENCY),
 CH_LAYOUT_4POINT0=           (CH_LAYOUT_SURROUND|CH_BACK_CENTER),
 CH_LAYOUT_4POINT1=           (CH_LAYOUT_4POINT0|CH_LOW_FREQUENCY),
 CH_LAYOUT_2_2=               (CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_SIDE_RIGHT),
 CH_LAYOUT_QUAD=              (CH_LAYOUT_STEREO|CH_BACK_LEFT|CH_BACK_RIGHT),
 CH_LAYOUT_5POINT0=           (CH_LAYOUT_SURROUND|CH_SIDE_LEFT|CH_SIDE_RIGHT),
 CH_LAYOUT_5POINT1=           (CH_LAYOUT_5POINT0|CH_LOW_FREQUENCY),
 CH_LAYOUT_5POINT0_BACK=      (CH_LAYOUT_SURROUND|CH_BACK_LEFT|CH_BACK_RIGHT),
 CH_LAYOUT_5POINT1_BACK=      (CH_LAYOUT_5POINT0_BACK|CH_LOW_FREQUENCY),
 CH_LAYOUT_6POINT0=           (CH_LAYOUT_5POINT0|CH_BACK_CENTER),
 CH_LAYOUT_6POINT0_FRONT=     (CH_LAYOUT_2_2|CH_FRONT_LEFT_OF_CENTER|CH_FRONT_RIGHT_OF_CENTER),
 CH_LAYOUT_HEXAGONAL=         (CH_LAYOUT_5POINT0_BACK|CH_BACK_CENTER),
 CH_LAYOUT_6POINT1=           (CH_LAYOUT_5POINT1|CH_BACK_CENTER),
 CH_LAYOUT_6POINT1_BACK=      (CH_LAYOUT_5POINT1_BACK|CH_BACK_CENTER),
 CH_LAYOUT_6POINT1_FRONT=     (CH_LAYOUT_6POINT0_FRONT|CH_LOW_FREQUENCY),
 CH_LAYOUT_7POINT0=           (CH_LAYOUT_5POINT0|CH_BACK_LEFT|CH_BACK_RIGHT),
 CH_LAYOUT_7POINT0_FRONT=     (CH_LAYOUT_5POINT0|CH_FRONT_LEFT_OF_CENTER|CH_FRONT_RIGHT_OF_CENTER),
 CH_LAYOUT_7POINT1=           (CH_LAYOUT_5POINT1|CH_BACK_LEFT|CH_BACK_RIGHT),
 CH_LAYOUT_7POINT1_WIDE=      (CH_LAYOUT_5POINT1|CH_FRONT_LEFT_OF_CENTER|CH_FRONT_RIGHT_OF_CENTER),
 CH_LAYOUT_7POINT1_WIDE_BACK= (CH_LAYOUT_5POINT1_BACK|CH_FRONT_LEFT_OF_CENTER|CH_FRONT_RIGHT_OF_CENTER),
 CH_LAYOUT_OCTAGONAL=         (CH_LAYOUT_5POINT0|CH_BACK_LEFT|CH_BACK_CENTER|CH_BACK_RIGHT),
 CH_LAYOUT_HEXADECAGONAL=     (CH_LAYOUT_OCTAGONAL|CH_WIDE_LEFT|CH_WIDE_RIGHT|CH_TOP_BACK_LEFT|CH_TOP_BACK_RIGHT|CH_TOP_BACK_CENTER|CH_TOP_FRONT_CENTER|CH_TOP_FRONT_LEFT|CH_TOP_FRONT_RIGHT),
 CH_LAYOUT_STEREO_DOWNMIX=    (CH_STEREO_LEFT|CH_STEREO_RIGHT),
  
     } ;

◆ Format

enum com::avpkit::core::IAudioSamples::Format

The format we use to represent audio.

Today only FMT_S16 (signed integer 16-bit audio) is supported.

Enumerator
FMT_U8	unsigned 8 bits
FMT_S16	signed 16 bits
FMT_S32	signed 32 bits
FMT_FLT	float
FMT_DBL	double
FMT_U8P	unsigned 8 bits, planar
FMT_S16P	signed 16 bits, planar
FMT_S32P	signed 32 bits, planar
FMT_FLTP	float, planar
FMT_DBLP	double, planar

Definition at line 46 of file IAudioSamples.h.

                         {
       FMT_NONE = -1,
       FMT_U8,          
       FMT_S16,         
       FMT_S32,         
       FMT_FLT,         
       FMT_DBL,         
  
       FMT_U8P,         
       FMT_S16P,        
       FMT_S32P,        
       FMT_FLTP,        
       FMT_DBLP,        
     } Format;

Member Function Documentation

◆ defaultPtsToSamples()

int64_t com::avpkit::core::IAudioSamples::defaultPtsToSamples	(	int64_t	duration,
		int	sampleRate
	)

static

Converts a duration in microseconds into a number of samples, assuming a given sampleRate.

Parameters

duration	The duration in microseconds.
sampleRate	sample rate that you want to use.

Returns: The number of samples it would take (at the given sampleRate) to take duration microseconds to play.

Definition at line 86 of file IAudioSamples.cpp.

   {
     int64_t retval = 0;
     Global::init();
     if (duration != Global::NO_PTS)
     {
       int64_t num = duration * sampleRate;
       int64_t den = Global::DEFAULT_PTS_PER_SECOND;
       long double calc = ((long double)num)/((long double)den);
       retval = (int64_t)(calc);
     }
     return retval;
   }

References com::avpkit::core::Global::DEFAULT_PTS_PER_SECOND, com::avpkit::core::Global::init(), and com::avpkit::core::Global::NO_PTS.

◆ findSampleBitDepth()

int32_t com::avpkit::core::IAudioSamples::findSampleBitDepth ( Format format )

static

A convenience method that returns the # of bits in a given format.

Be aware that right now this library only supports 16-bit audio.

Parameters

format The format you want to find the number of bits in.

Returns: The number of bits (not bytes) in the passed in format.

Definition at line 38 of file IAudioSamples.cpp.

   {
     int bits = av_get_bytes_per_sample((enum AVSampleFormat) format)<<3;
     return bits;
   }

Referenced by com::avpkit::core::AudioSamples::getSampleBitDepth(), and com::avpkit::core::AudioResampler::resample().

◆ getChannels()

virtual int32_t com::avpkit::core::IAudioSamples::getChannels ( )

pure virtual

Return the number of channels of the samples in this buffer.

For example, 1 is mono, 2 is stereo.

Returns: The number of channels.

Implemented in com::avpkit::core::AudioSamples.

Referenced by com::avpkit::core::AudioResampler::getMinimumNumSamplesRequiredInOutputSamples().

◆ getFormat()

virtual Format com::avpkit::core::IAudioSamples::getFormat ( )

pure virtual

Find the Format of the samples in this buffer.

Right now only FMT_S16 is supported.

Returns: The format of the samples.

Implemented in com::avpkit::core::AudioSamples.

Referenced by com::avpkit::core::AudioResampler::getMinimumNumSamplesRequiredInOutputSamples().

◆ getMaxBufferSize()

virtual int32_t com::avpkit::core::IAudioSamples::getMaxBufferSize ( )

pure virtual

Returns: Maximum number of bytes that can be put in this buffer. To get the number of samples you can put in this IAudioSamples instance, do the following num_samples = getMaxBufferSize() / (getSampleSize())

Implemented in com::avpkit::core::AudioSamples.

◆ getMaxSamples()

virtual int32_t com::avpkit::core::IAudioSamples::getMaxSamples ( )

pure virtual

Returns: Maximum number of samples this buffer can hold.

Implemented in com::avpkit::core::AudioSamples.

◆ getNextPts()

virtual int64_t com::avpkit::core::IAudioSamples::getNextPts ( )

pure virtual

What would be the next Presentation Time Stamp after all the samples in this buffer were played?

Returns: the next presentation time stamp (pts)

Implemented in com::avpkit::core::AudioSamples.

◆ getNumSamples()

virtual int32_t com::avpkit::core::IAudioSamples::getNumSamples ( )

pure virtual

Get the number of samples in this video.

For example, if you have 100 bytes of stereo (2-channel) 16-bit audio in this buffer, there are 25 samples. If you have 100 bytes of mono (1-channel) 16-bit audio in this buffer, you have 50 samples.

Returns: The number of samples.

Implemented in com::avpkit::core::AudioSamples.

Referenced by com::avpkit::core::AudioResampler::getMinimumNumSamplesRequiredInOutputSamples().

◆ getPts()

virtual int64_t com::avpkit::core::IAudioSamples::getPts ( )

pure virtual

What is the Presentation Time Stamp of this set of audio samples.

Returns: the presentation time stamp (pts)

Implemented in com::avpkit::core::AudioSamples.

◆ getSample()

virtual int32_t com::avpkit::core::IAudioSamples::getSample	(	int32_t	sampleIndex,
		int32_t	channel,
		Format	format
	)

pure virtual

Get the sample at the given sampleIndex and channel, and return it in the asked for format.

Parameters

sampleIndex	The zero-based index into this set of samples.
channel	The zero-based channel to get the sample from
format	The format to return in

Returns: The sample if available. If that sample is not available (e.g. because the channel doesn't exist, or the samples have not been setComplete(bool, int32_t, int32_t, int32_t, Format, int64_t)), then this method returns 0. It is up to the caller to ensure the inputs are valid given that 0 is also a valid sample value.

Implemented in com::avpkit::core::AudioSamples.

◆ getSampleBitDepth()

virtual int32_t com::avpkit::core::IAudioSamples::getSampleBitDepth ( )

pure virtual

Find out the bit-depth of the samples in this buffer.

Returns: Number of bits in a raw sample (per channel)

Implemented in com::avpkit::core::AudioSamples.

◆ getSampleRate()

virtual int32_t com::avpkit::core::IAudioSamples::getSampleRate ( )

pure virtual

Find the sample rate of the samples in this audio buffer.

Returns: The Sampling Rate of the samples in this buffer (e.g. 22050).

Implemented in com::avpkit::core::AudioSamples.

Referenced by com::avpkit::core::AudioResampler::getMinimumNumSamplesRequiredInOutputSamples().

◆ getSampleSize()

virtual int32_t com::avpkit::core::IAudioSamples::getSampleSize ( )

pure virtual

Returns: Number of bytes in a single sample of audio (including channels). You can also get this by getSampleBitDepth()*getChannels()/8.

Implemented in com::avpkit::core::AudioSamples.

◆ isComplete()

virtual bool com::avpkit::core::IAudioSamples::isComplete ( )

pure virtual

Returns whether or not we think this buffer has been filled with data.

Returns: Has setComplete() been called and the buffer populated.

Implemented in com::avpkit::core::AudioSamples.

Referenced by com::avpkit::core::AudioResampler::getMinimumNumSamplesRequiredInOutputSamples().

◆ make() [1/3]

IAudioSamples * com::avpkit::core::IAudioSamples::make	(	com::avpkit::ferry::IBuffer *	buffer,
		int	channels,
		IAudioSamples::Format	format
	)

static

Creates an IAudioSamples object by wrapping an object.

If you are decoding into this buffer, the buffer must be at least 192k*channels large (an FFmpeg requirement) or the decodeAudio call on IStreamCoder will fail with an error. If you are encoding from, any size should do.

Parameters

buffer	the buffer to wrap
channels	the number of channels of audio you will put it the buffer
format	the audio sample format

Returns: a new IAudioSamples object, or null on error.

Definition at line 61 of file IAudioSamples.cpp.

   {
     Global::init();
     return AudioSamples::make(buffer, channels, format);
   }

References com::avpkit::core::Global::init().

◆ make() [2/3]

IAudioSamples * com::avpkit::core::IAudioSamples::make	(	int32_t	numSamples,
		int32_t	numChannels
	)

static

Get a new audio samples buffer.

Note that any buffers this objects needs will be lazily allocated (i.e. we won't actually grab all the memory until we need it).

Parameters

numSamples	The minimum number of samples you're going to want to put in this buffer. We may (and probably will) return a larger buffer, but you cannot assume that.
numChannels	The number of channels in the audio you'll want to put in this buffer.

Returns: A new object, or null if we can't allocate one.

Definition at line 45 of file IAudioSamples.cpp.

   {
     Global::init();
     return AudioSamples::make(numSamples, numChannels);
   }

References com::avpkit::core::Global::init().

◆ make() [3/3]

IAudioSamples * com::avpkit::core::IAudioSamples::make	(	int32_t	numSamples,
		int32_t	numChannels,
		IAudioSamples::Format	format
	)

static

Get a new audio samples buffer.

Note that any buffers this objects needs will be lazily allocated (i.e. we won't actually grab all the memory until we need it).

Parameters

numSamples	The minimum number of samples you're going to want to put in this buffer. We may (and probably will) return a larger buffer, but you cannot assume that.
numChannels	The number of channels in the audio you'll want to put in this buffer.
format	The format of this buffer

Returns: A new object, or null if we can't allocate one.

Definition at line 52 of file IAudioSamples.cpp.

   {
     Global::init();
     return AudioSamples::make(numSamples, numChannels, format);
   }

References com::avpkit::core::Global::init().

◆ samplesToDefaultPts()

int64_t com::avpkit::core::IAudioSamples::samplesToDefaultPts	(	int64_t	samples,
		int	sampleRate
	)

static

Converts a number of samples at a given sampleRate into Microseconds.

Parameters

samples	Number of samples.
sampleRate	sample rate that those samples are recorded at.

Returns: number of microseconds it would take to play that audio.

Definition at line 69 of file IAudioSamples.cpp.

   {
     // Note: These need to always round up!  a "partial sample" actually must
     // be whole (and similar with time stamps).
     int64_t retval = Global::NO_PTS;
     Global::init();
     if (sampleRate > 0)
     {
       int64_t num = samples * Global::DEFAULT_PTS_PER_SECOND;
       int64_t den = sampleRate;
       long double calc = ((long double)num)/((long double)den);
       retval = (int64_t)(calc);
     }
     return retval;
   }

References com::avpkit::core::Global::DEFAULT_PTS_PER_SECOND, com::avpkit::core::Global::init(), and com::avpkit::core::Global::NO_PTS.

Referenced by com::avpkit::core::AudioSamples::getNextPts(), and com::avpkit::core::AudioResampler::resample().

◆ setComplete()

virtual void com::avpkit::core::IAudioSamples::setComplete	(	bool	complete,
		int32_t	numSamples,
		int32_t	sampleRate,
		int32_t	channels,
		Format	format,
		int64_t	pts
	)

pure virtual

Call this if you modify the samples and are now done.

This updates the pertinent information in the structure.

Parameters

complete	Is this set of samples complete?
numSamples	Number of samples in this update (note that 4 shorts of 16-bit audio in stereo is actually 1 sample).
sampleRate	The sample rate (in Hz) of this set of samples.
channels	The number of channels in this set of samples.
format	The sample-format of this set of samples.
pts	The presentation time stamp of the starting sample in this buffer. Caller must ensure pts is in units of 1/1,000,000 of a second

Implemented in com::avpkit::core::AudioSamples.

◆ setPts()

virtual void com::avpkit::core::IAudioSamples::setPts ( int64_t aValue )

pure virtual

Set the Presentation Time Stamp for this set of samples.

Parameters

aValue the new value

Implemented in com::avpkit::core::AudioSamples.

◆ setSample()

virtual int32_t com::avpkit::core::IAudioSamples::setSample	(	int32_t	sampleIndex,
		int32_t	channel,
		Format	format,
		int32_t	sample
	)

pure virtual

Sets the sample at the given index and channel to the sample.

In theory we assume input is the given Format, and will convert if needed, but right now we only support FMT_S16 anyway.

Parameters

sampleIndex	The zero-based index into the set of samples
channel	The zero-based channel number. If this set of samples doesn't have that given channel, an error is returned.
format	The format of the given sample
sample	The actual sample

Returns: >= 0 on success; -1 on error.

Implemented in com::avpkit::core::AudioSamples.

The documentation for this class was generated from the following files:

csrc/com/avpkit/core/IAudioSamples.h
csrc/com/avpkit/core/IAudioSamples.cpp

Public Types

Public Member Functions

Static Public Member Functions

Additional Inherited Members

Detailed Description

Member Typedef Documentation

◆ Format

Member Enumeration Documentation

◆ ChannelLayout

◆ Format

Member Function Documentation

◆ defaultPtsToSamples()

◆ findSampleBitDepth()

◆ getChannels()

◆ getFormat()

◆ getMaxBufferSize()

◆ getMaxSamples()

◆ getNextPts()

◆ getNumSamples()

◆ getPts()

◆ getSample()

◆ getSampleBitDepth()

◆ getSampleRate()

◆ getSampleSize()

◆ isComplete()

◆ make() [1/3]

◆ make() [2/3]

◆ make() [3/3]

◆ samplesToDefaultPts()

◆ setComplete()

◆ setPts()

◆ setSample()